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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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663 lines
17 KiB
C
663 lines
17 KiB
C
/* GStreamer
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* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
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* 2000 Wim Taymans <wtay@chello.be>
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* 2002 Kristian Rietveld <kris@gtk.org>
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* 2002,2003 Colin Walters <walters@gnu.org>
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* 2001,2010 Bastien Nocera <hadess@hadess.net>
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* 2010 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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*
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* rtmpsrc.c:
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-rtmpsrc
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*
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* This plugin reads data from a local or remote location specified
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* by an URI. This location can be specified using any protocol supported by
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* the RTMP library, i.e. rtmp, rtmpt, rtmps, rtmpe, rtmfp, rtmpte and rtmpts.
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*
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* <refsect2>
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* <title>Example launch lines</title>
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* |[
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* gst-launch-1.0 -v rtmpsrc location=rtmp://somehost/someurl ! fakesink
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* ]| Open an RTMP location and pass its content to fakesink.
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* </refsect2>
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <glib/gi18n-lib.h>
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#include "gstrtmpsrc.h"
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <gst/gst.h>
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#ifdef G_OS_WIN32
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#include <winsock2.h>
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#endif
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GST_DEBUG_CATEGORY_STATIC (rtmpsrc_debug);
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#define GST_CAT_DEFAULT rtmpsrc_debug
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static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS_ANY);
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enum
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{
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PROP_0,
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PROP_LOCATION,
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PROP_TIMEOUT
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#if 0
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PROP_SWF_URL,
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PROP_PAGE_URL
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#endif
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};
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#define DEFAULT_LOCATION NULL
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#define DEFAULT_TIMEOUT 120
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static void gst_rtmp_src_uri_handler_init (gpointer g_iface,
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gpointer iface_data);
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static void gst_rtmp_src_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_rtmp_src_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void gst_rtmp_src_finalize (GObject * object);
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static gboolean gst_rtmp_src_unlock (GstBaseSrc * src);
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static gboolean gst_rtmp_src_stop (GstBaseSrc * src);
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static gboolean gst_rtmp_src_start (GstBaseSrc * src);
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static gboolean gst_rtmp_src_is_seekable (GstBaseSrc * src);
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static gboolean gst_rtmp_src_prepare_seek_segment (GstBaseSrc * src,
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GstEvent * event, GstSegment * segment);
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static gboolean gst_rtmp_src_do_seek (GstBaseSrc * src, GstSegment * segment);
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static GstFlowReturn gst_rtmp_src_create (GstPushSrc * pushsrc,
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GstBuffer ** buffer);
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static gboolean gst_rtmp_src_query (GstBaseSrc * src, GstQuery * query);
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#define gst_rtmp_src_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstRTMPSrc, gst_rtmp_src, GST_TYPE_PUSH_SRC,
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G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER,
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gst_rtmp_src_uri_handler_init));
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static void
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gst_rtmp_src_class_init (GstRTMPSrcClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseSrcClass *gstbasesrc_class;
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GstPushSrcClass *gstpushsrc_class;
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gobject_class = G_OBJECT_CLASS (klass);
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gstelement_class = GST_ELEMENT_CLASS (klass);
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gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
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gstpushsrc_class = GST_PUSH_SRC_CLASS (klass);
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gobject_class->finalize = gst_rtmp_src_finalize;
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gobject_class->set_property = gst_rtmp_src_set_property;
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gobject_class->get_property = gst_rtmp_src_get_property;
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/* properties */
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g_object_class_install_property (gobject_class, PROP_LOCATION,
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g_param_spec_string ("location", "RTMP Location",
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"Location of the RTMP url to read",
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DEFAULT_LOCATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_TIMEOUT,
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g_param_spec_int ("timeout", "RTMP Timeout",
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"Time without receiving any data from the server to wait before to timeout the session",
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0, G_MAXINT,
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DEFAULT_TIMEOUT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gst_element_class_add_static_pad_template (gstelement_class, &srctemplate);
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gst_element_class_set_static_metadata (gstelement_class,
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"RTMP Source",
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"Source/File",
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"Read RTMP streams",
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"Bastien Nocera <hadess@hadess.net>, "
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"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
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gstbasesrc_class->start = GST_DEBUG_FUNCPTR (gst_rtmp_src_start);
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gstbasesrc_class->stop = GST_DEBUG_FUNCPTR (gst_rtmp_src_stop);
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gstbasesrc_class->unlock = GST_DEBUG_FUNCPTR (gst_rtmp_src_unlock);
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gstbasesrc_class->is_seekable = GST_DEBUG_FUNCPTR (gst_rtmp_src_is_seekable);
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gstbasesrc_class->prepare_seek_segment =
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GST_DEBUG_FUNCPTR (gst_rtmp_src_prepare_seek_segment);
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gstbasesrc_class->do_seek = GST_DEBUG_FUNCPTR (gst_rtmp_src_do_seek);
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gstpushsrc_class->create = GST_DEBUG_FUNCPTR (gst_rtmp_src_create);
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gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_rtmp_src_query);
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GST_DEBUG_CATEGORY_INIT (rtmpsrc_debug, "rtmpsrc", 0, "RTMP Source");
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}
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static void
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gst_rtmp_src_init (GstRTMPSrc * rtmpsrc)
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{
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#ifdef G_OS_WIN32
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WSADATA wsa_data;
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if (WSAStartup (MAKEWORD (2, 2), &wsa_data) != 0) {
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GST_ERROR_OBJECT (rtmpsrc, "WSAStartup failed: 0x%08x", WSAGetLastError ());
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}
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#endif
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rtmpsrc->cur_offset = 0;
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rtmpsrc->last_timestamp = 0;
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rtmpsrc->timeout = DEFAULT_TIMEOUT;
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gst_base_src_set_format (GST_BASE_SRC (rtmpsrc), GST_FORMAT_TIME);
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}
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static void
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gst_rtmp_src_finalize (GObject * object)
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{
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GstRTMPSrc *rtmpsrc = GST_RTMP_SRC (object);
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g_free (rtmpsrc->uri);
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rtmpsrc->uri = NULL;
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#ifdef G_OS_WIN32
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WSACleanup ();
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#endif
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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/*
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* URI interface support.
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*/
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static GstURIType
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gst_rtmp_src_uri_get_type (GType type)
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{
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return GST_URI_SRC;
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}
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static const gchar *const *
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gst_rtmp_src_uri_get_protocols (GType type)
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{
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static const gchar *protocols[] =
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{ "rtmp", "rtmpt", "rtmps", "rtmpe", "rtmfp", "rtmpte", "rtmpts", NULL };
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return protocols;
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}
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static gchar *
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gst_rtmp_src_uri_get_uri (GstURIHandler * handler)
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{
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GstRTMPSrc *src = GST_RTMP_SRC (handler);
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/* FIXME: make thread-safe */
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return g_strdup (src->uri);
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}
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static gboolean
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gst_rtmp_src_uri_set_uri (GstURIHandler * handler, const gchar * uri,
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GError ** error)
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{
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GstRTMPSrc *src = GST_RTMP_SRC (handler);
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if (GST_STATE (src) >= GST_STATE_PAUSED) {
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g_set_error (error, GST_URI_ERROR, GST_URI_ERROR_BAD_STATE,
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"Changing the URI on rtmpsrc when it is running is not supported");
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return FALSE;
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}
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g_free (src->uri);
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src->uri = NULL;
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if (uri != NULL) {
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int protocol;
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AVal host;
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unsigned int port;
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AVal playpath, app;
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if (!RTMP_ParseURL (uri, &protocol, &host, &port, &playpath, &app) ||
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!host.av_len || !playpath.av_len) {
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GST_ERROR_OBJECT (src, "Failed to parse URI %s", uri);
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g_set_error (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
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"Could not parse RTMP URI");
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/* FIXME: we should not be freeing RTMP internals to avoid leaking */
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free (playpath.av_val);
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return FALSE;
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}
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free (playpath.av_val);
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src->uri = g_strdup (uri);
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}
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GST_DEBUG_OBJECT (src, "Changed URI to %s", GST_STR_NULL (uri));
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return TRUE;
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}
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static void
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gst_rtmp_src_uri_handler_init (gpointer g_iface, gpointer iface_data)
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{
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GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
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iface->get_type = gst_rtmp_src_uri_get_type;
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iface->get_protocols = gst_rtmp_src_uri_get_protocols;
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iface->get_uri = gst_rtmp_src_uri_get_uri;
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iface->set_uri = gst_rtmp_src_uri_set_uri;
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}
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static void
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gst_rtmp_src_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstRTMPSrc *src;
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src = GST_RTMP_SRC (object);
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switch (prop_id) {
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case PROP_LOCATION:{
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gst_rtmp_src_uri_set_uri (GST_URI_HANDLER (src),
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g_value_get_string (value), NULL);
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break;
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}
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case PROP_TIMEOUT:{
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src->timeout = g_value_get_int (value);
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break;
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}
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_rtmp_src_get_property (GObject * object, guint prop_id, GValue * value,
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GParamSpec * pspec)
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{
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GstRTMPSrc *src;
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src = GST_RTMP_SRC (object);
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switch (prop_id) {
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case PROP_LOCATION:
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g_value_set_string (value, src->uri);
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break;
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case PROP_TIMEOUT:
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g_value_set_int (value, src->timeout);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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/*
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* Read a new buffer from src->reqoffset, takes care of events
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* and seeking and such.
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*/
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static GstFlowReturn
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gst_rtmp_src_create (GstPushSrc * pushsrc, GstBuffer ** buffer)
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{
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GstRTMPSrc *src;
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GstBuffer *buf;
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GstMapInfo map;
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guint8 *data;
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guint todo;
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gsize bsize;
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int read;
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int size;
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src = GST_RTMP_SRC (pushsrc);
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g_return_val_if_fail (src->rtmp != NULL, GST_FLOW_ERROR);
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size = GST_BASE_SRC_CAST (pushsrc)->blocksize;
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GST_DEBUG ("reading from %" G_GUINT64_FORMAT
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", size %u", src->cur_offset, size);
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buf = gst_buffer_new_allocate (NULL, size, NULL);
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if (G_UNLIKELY (buf == NULL)) {
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GST_ERROR_OBJECT (src, "Failed to allocate %u bytes", size);
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return GST_FLOW_ERROR;
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}
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bsize = todo = size;
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gst_buffer_map (buf, &map, GST_MAP_WRITE);
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data = map.data;
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read = bsize = 0;
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while (todo > 0) {
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read = RTMP_Read (src->rtmp, (char *) data, todo);
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if (G_UNLIKELY (read == 0 && todo == size)) {
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goto eos;
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} else if (G_UNLIKELY (read == 0)) {
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todo = 0;
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break;
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}
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if (G_UNLIKELY (read < 0))
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goto read_failed;
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if (read < todo) {
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data += read;
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todo -= read;
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bsize += read;
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} else {
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bsize += todo;
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todo = 0;
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}
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GST_LOG (" got size %d", read);
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}
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gst_buffer_unmap (buf, &map);
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gst_buffer_resize (buf, 0, bsize);
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if (src->discont) {
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GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
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src->discont = FALSE;
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}
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GST_BUFFER_TIMESTAMP (buf) = src->last_timestamp;
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GST_BUFFER_OFFSET (buf) = src->cur_offset;
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src->cur_offset += size;
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if (src->last_timestamp == GST_CLOCK_TIME_NONE)
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src->last_timestamp = src->rtmp->m_mediaStamp * GST_MSECOND;
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else
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src->last_timestamp =
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MAX (src->last_timestamp, src->rtmp->m_mediaStamp * GST_MSECOND);
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GST_LOG_OBJECT (src, "Created buffer of size %u at %" G_GINT64_FORMAT
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" with timestamp %" GST_TIME_FORMAT, size, GST_BUFFER_OFFSET (buf),
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GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
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/* we're done, return the buffer */
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*buffer = buf;
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return GST_FLOW_OK;
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read_failed:
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{
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gst_buffer_unref (buf);
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GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL), ("Failed to read data"));
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return GST_FLOW_ERROR;
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}
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eos:
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{
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gst_buffer_unref (buf);
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if (src->cur_offset == 0) {
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GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
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("Failed to read any data from stream, check your URL"));
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return GST_FLOW_ERROR;
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} else {
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GST_DEBUG_OBJECT (src, "Reading data gave EOS");
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return GST_FLOW_EOS;
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}
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}
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}
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static gboolean
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gst_rtmp_src_query (GstBaseSrc * basesrc, GstQuery * query)
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{
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gboolean ret = FALSE;
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GstRTMPSrc *src = GST_RTMP_SRC (basesrc);
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switch (GST_QUERY_TYPE (query)) {
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case GST_QUERY_URI:
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gst_query_set_uri (query, src->uri);
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ret = TRUE;
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break;
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case GST_QUERY_POSITION:{
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GstFormat format;
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gst_query_parse_position (query, &format, NULL);
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if (format == GST_FORMAT_TIME) {
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gst_query_set_position (query, format, src->last_timestamp);
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ret = TRUE;
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}
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break;
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}
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case GST_QUERY_DURATION:{
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GstFormat format;
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gdouble duration;
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gst_query_parse_duration (query, &format, NULL);
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if (format == GST_FORMAT_TIME && src->rtmp) {
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duration = RTMP_GetDuration (src->rtmp);
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if (duration != 0.0) {
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gst_query_set_duration (query, format, duration * GST_SECOND);
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ret = TRUE;
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}
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}
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break;
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}
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case GST_QUERY_SCHEDULING:{
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gst_query_set_scheduling (query,
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GST_SCHEDULING_FLAG_SEQUENTIAL |
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GST_SCHEDULING_FLAG_BANDWIDTH_LIMITED, 1, -1, 0);
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gst_query_add_scheduling_mode (query, GST_PAD_MODE_PUSH);
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ret = TRUE;
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break;
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}
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default:
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ret = FALSE;
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break;
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}
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if (!ret)
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ret = GST_BASE_SRC_CLASS (parent_class)->query (basesrc, query);
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return ret;
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}
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static gboolean
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gst_rtmp_src_is_seekable (GstBaseSrc * basesrc)
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{
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GstRTMPSrc *src;
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src = GST_RTMP_SRC (basesrc);
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return src->seekable;
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}
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static gboolean
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gst_rtmp_src_prepare_seek_segment (GstBaseSrc * basesrc, GstEvent * event,
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GstSegment * segment)
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{
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GstRTMPSrc *src;
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GstSeekType cur_type, stop_type;
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gint64 cur, stop;
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GstSeekFlags flags;
|
|
GstFormat format;
|
|
gdouble rate;
|
|
|
|
src = GST_RTMP_SRC (basesrc);
|
|
|
|
gst_event_parse_seek (event, &rate, &format, &flags,
|
|
&cur_type, &cur, &stop_type, &stop);
|
|
|
|
if (!src->seekable) {
|
|
GST_LOG_OBJECT (src, "Not a seekable stream");
|
|
return FALSE;
|
|
}
|
|
|
|
if (!src->rtmp) {
|
|
GST_LOG_OBJECT (src, "Not connected yet");
|
|
return FALSE;
|
|
}
|
|
|
|
if (format != GST_FORMAT_TIME) {
|
|
GST_LOG_OBJECT (src, "Seeking only supported in TIME format");
|
|
return FALSE;
|
|
}
|
|
|
|
if (stop_type != GST_SEEK_TYPE_NONE) {
|
|
GST_LOG_OBJECT (src, "Setting a stop position is not supported");
|
|
return FALSE;
|
|
}
|
|
|
|
gst_segment_init (segment, GST_FORMAT_TIME);
|
|
gst_segment_do_seek (segment, rate, format, flags, cur_type, cur, stop_type,
|
|
stop, NULL);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtmp_src_do_seek (GstBaseSrc * basesrc, GstSegment * segment)
|
|
{
|
|
GstRTMPSrc *src;
|
|
|
|
src = GST_RTMP_SRC (basesrc);
|
|
|
|
if (segment->format != GST_FORMAT_TIME) {
|
|
GST_LOG_OBJECT (src, "Only time based seeks are supported");
|
|
return FALSE;
|
|
}
|
|
|
|
if (!src->rtmp) {
|
|
GST_LOG_OBJECT (src, "Not connected yet");
|
|
return FALSE;
|
|
}
|
|
|
|
src->discont = TRUE;
|
|
|
|
/* Initial seek */
|
|
if (src->cur_offset == 0 && segment->start == 0)
|
|
return TRUE;
|
|
|
|
if (!src->seekable) {
|
|
GST_LOG_OBJECT (src, "Not a seekable stream");
|
|
return FALSE;
|
|
}
|
|
|
|
src->last_timestamp = GST_CLOCK_TIME_NONE;
|
|
if (!RTMP_SendSeek (src->rtmp, segment->start / GST_MSECOND)) {
|
|
GST_ERROR_OBJECT (src, "Seeking failed");
|
|
src->seekable = FALSE;
|
|
return FALSE;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (src, "Seek to %" GST_TIME_FORMAT " successfull",
|
|
GST_TIME_ARGS (segment->start));
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
#define STR2AVAL(av,str) G_STMT_START { \
|
|
av.av_val = str; \
|
|
av.av_len = strlen(av.av_val); \
|
|
} G_STMT_END;
|
|
|
|
/* open the file, do stuff necessary to go to PAUSED state */
|
|
static gboolean
|
|
gst_rtmp_src_start (GstBaseSrc * basesrc)
|
|
{
|
|
GstRTMPSrc *src;
|
|
|
|
src = GST_RTMP_SRC (basesrc);
|
|
|
|
if (!src->uri) {
|
|
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL), ("No filename given"));
|
|
return FALSE;
|
|
}
|
|
|
|
src->cur_offset = 0;
|
|
src->last_timestamp = 0;
|
|
src->discont = TRUE;
|
|
|
|
src->rtmp = RTMP_Alloc ();
|
|
if (!src->rtmp) {
|
|
GST_ERROR_OBJECT (src, "Could not allocate librtmp's RTMP context");
|
|
goto error;
|
|
}
|
|
|
|
RTMP_Init (src->rtmp);
|
|
src->rtmp->Link.timeout = src->timeout;
|
|
if (!RTMP_SetupURL (src->rtmp, src->uri)) {
|
|
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
|
|
("Failed to setup URL '%s'", src->uri));
|
|
goto error;
|
|
}
|
|
src->seekable = !(src->rtmp->Link.lFlags & RTMP_LF_LIVE);
|
|
GST_INFO_OBJECT (src, "seekable %d", src->seekable);
|
|
|
|
/* open if required */
|
|
if (!RTMP_IsConnected (src->rtmp)) {
|
|
if (!RTMP_Connect (src->rtmp, NULL)) {
|
|
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
|
|
("Could not connect to RTMP stream \"%s\" for reading", src->uri));
|
|
goto error;
|
|
}
|
|
}
|
|
|
|
return TRUE;
|
|
|
|
error:
|
|
if (src->rtmp) {
|
|
RTMP_Free (src->rtmp);
|
|
src->rtmp = NULL;
|
|
}
|
|
return FALSE;
|
|
}
|
|
|
|
#undef STR2AVAL
|
|
|
|
static gboolean
|
|
gst_rtmp_src_unlock (GstBaseSrc * basesrc)
|
|
{
|
|
GstRTMPSrc *rtmpsrc = GST_RTMP_SRC (basesrc);
|
|
|
|
GST_DEBUG_OBJECT (rtmpsrc, "unlock");
|
|
|
|
/* This closes the socket, which means that any pending socket calls
|
|
* error out. */
|
|
if (rtmpsrc->rtmp) {
|
|
RTMP_Close (rtmpsrc->rtmp);
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
|
|
static gboolean
|
|
gst_rtmp_src_stop (GstBaseSrc * basesrc)
|
|
{
|
|
GstRTMPSrc *src;
|
|
|
|
src = GST_RTMP_SRC (basesrc);
|
|
|
|
if (src->rtmp) {
|
|
RTMP_Free (src->rtmp);
|
|
src->rtmp = NULL;
|
|
}
|
|
|
|
src->cur_offset = 0;
|
|
src->last_timestamp = 0;
|
|
src->discont = TRUE;
|
|
|
|
return TRUE;
|
|
}
|